System and method for estimating initial channel quality in a multirate system

ABSTRACT

A method of estimating initial channel quality in a receiver when allocated a new channel in order to select an optimal codec mode in a multi-rate service is disclosed. One implementation initially fills the filter state with the first received channel quality measurement. Another embodiment proportionally fills the entire filter state with the calculated channel quality measurements. Yet another embodiment uses the hysteresis and threshold parameters in conjunction with the initial codec mode to calculate an initial fill value for the filter state.

BACKGROUND OF INVENTION

The following acronyms are used throughout this description. They arelisted in TABLE 1 below for ease of reference.

TABLE 1 ACRONYM DEFINITION ACELP Algebraic Code Excited LinearPrediction ACS Active Code Set AFS AMR Full Rate Speech service AHS AMRHalf Rate Speech Service AMR Adaptive Multi Rate Speech Service BER BitError Rate BSS Base Station Subsystem BTS Base Transceiver Station CDMACode Division Multiple Access CELP Code Excited Linear Prediction CHDChannel Decoder CHE Channel Encoder C/I Carrier-to-Interface Ratio (usedto measure link quality) CMI Codec Mode Indication CMC Codec ModeCommand CMR Coec Mode Request Codec 1 Lowest Speech bitrate mode in ACS(CMC/CMR/CMI = 0) Codec 2 2^(nd) lowest bitrate codec if ACS contains 2or more codecs (CMC/CMR/CMI = 1) Codec 3 3^(rd) lowest speech bitratecodec if ACS contains 3 or more codecs (CMC/CMR/CMI = 2) Codec 4 Highestspeech bitrate codec if ACS contains 4 codecs (CMC/CMR/CMI = 3) DLDownlink EDGE Enhanced Data-rates for GSM (or Global) Evolution EFREnhanced Full Rate Speech Codec for GSM FACCH Fast Associated ControlChannel FIR Finite Impulse Response FR Full Rate Speech Codec for GSMGSM Global System for Mobile Communications, Common Digital CellularStandard HR Half Rate Speech Codec for GSM ICM Initial Codec Mode IIRInfinite Impulse Response MS Mobile Station, e.g. a cellular phone PDCPersonal Digital Cellular RATSCCH Robust AMR Traffic SynchronizedControl Channel SACCH Slow Associated Control Channel SPD Speech DecoderSPE Speech Encoder TDMA Time Division Multiple Access TCH TrafficChannel TRAU Transcoding and Rate Adapting Unit UL Uplink WCDMAWideband-CDMA 3GPP 3^(rd) Generation Pasrtnership Project, WCDMAStandard

Digital communications systems, such as digital cellular telephonysystems, are often used to transmit voice. Due to the limited bandwidthof these systems, speech is typically encoded to a low bit rate using aspeech encoder. Various methods are in use for such speech coding.Within modern digital cellular telephony, most of these methods arebased upon Code Excited Linear Prediction (CELP) or some variantthereof. Such speech codecs are standardized and in use for all of themajor digital cellular telephony standards including GSM/EDGE, PDC,TDMA, CDMA, and WCDMA.

The present invention is described within the context of GSM. Withinthis standard, there are currently four standardized speech services;three of which are fielded and in common use. The original speech codecis known as the full-rate (FR) codec. This was followed by the half-rate(HR) speech codec which required only half of the bandwidth of the FRcodec thereby allowing cellular operators to support twice as many userswithin the same frequency allocation. This was followed by the EnhancedFull Rate (EFR) speech codec which required the same net bit rate (afterchannel coding) as the original FR codec but with much improved speechquality.

The GSM standard recently introduced the AMR speech codec. This speechcodec will also be used in forthcoming EDGE and 3GPP cellular systems.

Basic AMR Operation—The AMR speech service is fundamentally differentfrom fixed-rate codecs in that multiple speech rates are defined and thespeech rate may be changed dynamically. For each speech rate, a channelcoding scheme is defined. The speech rate may be altered in order tomatch the channel coding to the link quality. There is both a half-rateand full-rate mode of AMR operation.

The AMR speech service has capacity and speech quality advantages overpreceding GSM speech codecs. The capacity increase is largely aconsequence of there existing a half-rate option in addition to the fullrate. Because the ACELP technology underlying the AMR codec is superiorto that of the original HR speech codec, the AMR half rate codec willlikely be acceptable with respect to speech quality (unlike its HRpredecessor).

The adaptive nature of the AMR codec also serves to increase capacity.Because the channel coding can be increased dynamically as needed,additional coding gain is possible meaning that acceptable operation ata lower C/I ratio is possible. Because GSM systems are typicallyinterference-limited (as compared to noise-limited), a lower average C/Iratio effectively means it is possible to put more users on the samesystem.

Considering only the full-rate AMR service, the speech quality perceivedby the user is improved over the fixed-rate speech codecs (FR, EFR). Asan AMR user encounters poor channel conditions, e.g. at the edge of acell or within a fade, the speech rate is reduced and the channel codingincreased. The reduced speech rate leads to a lower but acceptablespeech quality. This dynamic change is performed in a controlled mannersuch that the BER into the speech decoder is always kept at acceptablelevels. With a fixed-rate speech codec under similar conditions, the biterror rate (BER) into the speech decoder would quickly reachunacceptable levels leading to degraded speech quality out of thedecoder.

A high-level block diagram of a complete AMR system is illustrated inFIG. 1. The AMR system is incorporated within the major GSM base stationcomponents TRAU and BTS on the network side and the MS or mobileterminal on the portable equipment side.

On the network side, the speech encoder (SPE) and channel encoder (CHE)as well as channel decoder (CHD) and speech decoder (SPD) are connectedvia the serial A-bis interface. For each link, quality information isderived by estimating the current channel quality. Based on the channelquality, and also taking into consideration possible constraints fromnetwork control, the codec mode control, which is located on the networkside, selects the codec modes to be applied.

The channel mode to use (TCH/AFS or TCH/AHS) is controlled by thenetwork. Uplink and downlink always apply the same channel mode.

In order to achieve the benefits of having multiple codec rates andcodec schemes, it is important to choose the optimal codec rate for agiven link (channel condition) at a given time. This is accomplished viaa closed loop control mechanism wherein the network actually controlsthe rates on both links based on a channel quality measure.

For codec mode adaptation the receiving side performs link qualitymeasurements of the incoming link. The measurements are processedyielding a Quality Indicator. For uplink adaptation, the QualityIndicator is directly fed into the UL mode control unit. This unitcompares the Quality Indicator with certain thresholds and generates,also considering possible constraints from network control, a Codec ModeCommand indicating the codec mode to be used on the uplink. The CodecMode Command is then transmitted inband to the mobile side where theincoming speech signal is encoded in the corresponding codec mode. Fordownlink adaptation, the DL Mode Request Generator within the mobilecompares the DL Quality indicator with certain thresholds and generatesa Codec Mode Request indicating the preferred codec mode for thedownlink. The Codec Mode Request is transmitted inband to the networkside where it is fed into the DL Mode Control unit. This unit generallygrants the requested mode. However, considering possible constraintsfrom network control, it may also override the request. The resultingcodec mode is then applied for encoding of the incoming speech signal indownlink direction. Both for uplink and downlink, the presently appliedcodec mode is transmitted inband as Codec Mode Indication together withthe coded speech data. At the decoder, the Codec Mode Indication isdecoded and applied for decoding of the received speech data.

Codec mode selection is chosen from a set of codec modes (ACS, ActiveCodec Set), which may include 1 to 4 AMR codec modes. For the full-ratechannel, a total of 8 speech codec rates are defined and each has aunique channel code associated with it. For the half-rate service, fewerspeech rates are used. The ACS represents a subset of these codec rates.Associated with this set is a list of 0 to 3 switching thresholds andhystereses used by the DL Mode Request Generator and the UL mode controlunit to generate the Codec Mode Requests and Codec Mode Commands. Theseconfiguration parameters (ACS, thresholds, hysteresis) are defined atcall set-up and can be modified at handover or during a call, e.g. viaspecial reconfiguration messages sent on the RATSCCH channel. Inaddition, separate configuration parameters may be defined for uplinkand downlink channels.

Transmittal of the codec mode is done via special inband signaling. Thisinband signaling is comprised of a 2-bit index into the ACS. The valueof interest is channel encoded to 4 bits for AHS and 8 bits for AFS.These bits are allocated within the 456-bit AFS speech frame or the228-bit AHS speech frame leaving 448 bits and 224 bits, respectively,for the encoded speech frame.

Each coded frame contains only one inband data value and the inband dataalternates in meaning. For every other speech frame, the inband datarepresents the Codec Mode Indication (CMI). The CMI indicates the codecmode used on the link in which it is being transmitted and, hence, howreceived speech frames should be decoded, i.e. which channel coding todecode. Each CMI must be statically stored in the receiver as itindicates how to decode both the frame in which it was received and thenext frame.

The inband data in the remaining frames gives instructions concerningthe mode that should be used on the opposite link. For the data in thedownlink frames, this field is termed the codec mode command (CMC) asthis value commands the MS as to which codec mode to use on the uplink.For the data in the uplink frames, this field is termed the codec moderequest (CMR) as this is the codec mode which the MS would like to seeused on the downlink. The network will typically use the CMR to drivethe downlink codec rate but it is not required to do so.

Given the aforementioned link adaptation, it remains to determine howthe optimal codec mode is chosen. This is accomplished by estimating thechannel quality for a given link and using this to determine the optimalcodec rate from those within the ACS. For simplicity, henceforth onlythe downlink to the MS receiver will be considered. It is understoodthat the concepts apply equally to the uplink (BTS receiver).

The channel quality must be monitored by the MS receiver. A GSM receiveralready monitors this link and reports RXLEV (received signal strength)and RXQUAL (received signal quality) parameters back to the BTS viaSACCH. However, these methods for measuring channel quality aregenerally not adequate for purposes of AMR link adaptation.

The channel quality for AMR is mapped to a Carrier-to-Interference (C/I)ratio between 0 and 31.5 dB, though the practical range of interest istypically narrower. Multiple methods for estimating this channel qualitymeasurement are possible. The most obvious method is to explicitlyestimate the carrier and interferer energies. Other measurementtechniques such as estimating BER and mapping it to a C/I value or usingViterbi metrics are possible. The particular method for channel qualityestimation is outside of the scope of this disclosure and thus any ofthe methods are applicable. The only presumption herein is that thechannel quality measurement be estimated on each received burst or frameand that it be mappable to a C/I ratio.

The C/I ratio is compared against certain thresholds to calculate theCMR value to be returned to the BTS. In general, a strong signal (C/Iratio) indicates that a high speech rate (low channel coding) may beused on the downlink. A weak signal indicates more channel coding (and alow speech rate) are needed.

The exact thresholds are signaled by the network during AMRconfiguration, i.e. at the same time as the ACS is defined. In order toprevent the codec mode from oscillating excessively between two rateswhen the channel quality is near a threshold, hysteresis values are alsopart of the AMR configuration. In a typical AMR receiver, input burstsof RF data are demodulated and assembled into frames. These are passedto a channel decoder whose results are sent to the speech decoder. Inparallel with this, a channel quality measurement is performed basedupon inputs from the raw bursts, the demodulator, the channel decoder,and/or the channel decoder input/output data. The channel quality isused to calculate a mode request value.

The channel quality (C/I estimate) is typically measured on a burst(TDMA frame) or speech frame basis. (In AFS, a speech frame is composedof 4 bursts whereas in AHS 2 bursts comprise a speech frame.)Considerable variance will be observed in these measurements due to fastfades, antenna orientation, and other fleeting channel perturbations.Frequency hopping (typical in GSM systems) adds an additionalfast-varying source of channel condition changes.

It is not practical to track the fast-varying channel conditions as thiswould effectively require some prediction that is unfeasible given therandomness. Hence, it is desirable to remove the fast-varying componentsand concentrate on the long-term average (slow-varying) channel quality.This is achieved by filtering the channel quality estimates measured ona burst or frame basis.

The Link Adaptation standard suggests that a 500 ms (100-tap for AFSbursts, 50-tap for AHS bursts) FIR (moving average) filter be used. Forpurposes of this disclosure, it is assumed such a filter would be used.It is understood that an alternate filter including a different numberof FIR taps and/or poles (IIR filter) may be similarly used within theinvention.

When a traffic channel is assigned at the beginning of a call or a newtraffic channel is assigned as the result of a handover, the mobileusually begins using it with no prior knowledge of channel quality.Since a filter relies on past values, how the designer chooses toinitialize the C/I estimate filter state directly affects initialperformance. The simplest approach is to initialize the filter statewith a fixed value such as zero. Actual C/I measurements displace thepreloaded initial value over time. Given the situation where the ICM isnon-optimal for the current RF conditions, the request for the optimalcodec mode is delayed until the filter state is dominated by actualmeasurements.

Consider the case where the mobile is ordered to begin a call using itslowest bit rate codec (this is very common when using implicit rules forstartmode) yet the RF channel quality is capable of supporting thehighest bit rate codec. The mobile stays at the lower rate codec untilenough zeros are displaced that the filtered C/I estimate exceeds thethreshold required to request the next highest rate codec in the ACS.This process continues until the highest rate codec is requested. Duringthis process, the user experiences degraded audio until the optimalcodec is requested.

Another problem arises when the base station orders the highest-ratecodec to be used initially. With the filter state initialized withzeros, it is likely that the mobile will request a lower bit rate codecat the call start based on the first calculation from the filter. Theoptimal codec is requested only after actual RF measurements dominatethe initialization value in the C/I estimate filter. To prevent thisstartup problem, the ability to request a new codec could be barreduntil the filter is filled with actual RF measurements. Again thisarbitrary implementation results in unnecessarily degraded audio at thebeginning of a call (or immediately after handover) even though theradio link quality is capable of supporting the highest bit rate speechcodec. Although these preceding examples focus on C/I estimate filtersinitialized with zero, one skilled in the art recognizes that similarproblems exist with any constant value.

The ICM is typically not chosen by the base station arbitrarily ateither call setup or handover. The base station is familiar with networkaspects that affect channel quality such as the cell layout and the cellloading. A lightly loaded cell in a network having small cells will tendto have good channel quality even at handover. Conversely, aheavily-loaded network or one having large cells will tend to have poorchannel quality at a handover. Furthermore, a network usually has someexplicit indication of the channel quality as reported by RXQUAL orRXLEV measurements taken by the MS as part of the mobile-assistedhandover procedure when changing TCH channels or as part of the SDCCHmeasurements taken before switching over to the TCH at call setup.Concerning handovers, the network typically requires that the RXLEVreported by the MS for a neighbor cell is better than that of theserving cell before the handover is commanded. Hence, the network alwayshas a rough estimate of the MS's received channel quality.

Given the above, the network is in a better position to determine theICM that should be used for the downlink to the MS at the beginning of acall or at handover. Hence, the commanded ICM value should be used untilthe MS can estimate the channel quality with a reasonable degree ofconfidence, e.g. when the filter is filled with measured rather thaninitial values.

The issue with the previously described filter initialization is that iteffectively overrides the intent of the network when a new channel isassigned. The network is, in this respect, smarter than the MS and in abetter position to choose the AMR rate. If the MS initializes to a setvalue (such as zero), the CMR is forced to a non-optimal codec (thelowest) until the filter is filled with actual measurements.

The problem of a non-optimal codec is probably not too significant atcall setup. However, switching to a non-optimal codec at each handoveris quite undesirable. Note that a primary scenario (and motivation) forAMR operation is in weak signal conditions, e.g. at cell boundaries. Atcell boundaries (e.g., marginal channel quality for large cells or goodchannel quality in small cells) multiple handovers may occur leading tosignificant periods of time in which such a non-optimal codec would beused. The problem is further exacerbated if a high-order FIR filter isused. A higher-order filter would effectively lengthen the periods oftime during which a non-optimal codec is used.

What is desired is a method for determining initial channel qualityestimates that would allow an optimal codec to be used as often as ispractical whenever a new RF channel is allocated.

SUMMARY OF INVENTION

The present invention comprises a method of estimating initial RFchannel quality in a receiver when using a new RF channel in order toselect an optimal codec mode in a multi-rate service. A new RF channelis typically received at call setup or due to a handover.

In one embodiment of the present invention there is disclosed a methodof estimating initial RF channel quality in a receiver when receiving anew RF channel in order to select an optimal codec mode in a multi-ratespeech service. Channel quality estimates are obtained on a burst orframe basis. A carrier-to-interference (C/I) ratio is then calculatedfor each incoming burst or frame. A filter comprised of N taps isdynamically initialized with the calculated C/I ratios. In thisembodiment, dynamically initializing the filter comprises initializingall N taps of the filter with the first calculated C/I ratio. Finally,the C/I ratios are filtered to remove randomness. The result is a betterchannel quality estimation that can be used in codec mode selectionwithin the multi-rate speech service.

In another embodiment, dynamically initializing the filter with thecalculated C/I ratios comprises proportionally filling the N taps witheach calculated C/I ratio such that each calculated C/I ratio isproportionally represented in the filter. Remaining taps due to unevendivisions are filled with the most recent calculated C/I ratio. Thismethod applies until the number of calculated C/I ratios equals ½ thetotal number of filter taps. New calculated C/I ratios are then inputnormally, i.e. on a one in/one out basis.

In yet another embodiment, dynamically initializing the filter with thecalculated C/I ratios comprises obtaining the number of codec modes forthe multi-rate speech service, threshold and hysteresis values for theACS, and the initial codec mode from a base station that is incommunication with the receiver. Next, a value to be used to initializeall N taps of the filter based upon the threshold and hysteresis valuesof the codec modes is calculated by determining the average value of theinitial codec mode using the appropriate threshold and hysteresis valuesobtained from the base station. Lower practical limits are used when thecodec mode is the lowest bit rate codec mode. Similarly, upper practicallimits are used when the codec mode is the highest bit rate codec mode.

In still another embodiment, dynamically initializing the filter withthe calculated C/I ratios comprises obtaining the number of codec modesfor the multi-rate speech service, threshold and hysteresis values foreach codec mode, and the initial codec mode from a base station that isin communication with the receiver. Next, a value to be used toinitialize all N taps of the filter based upon the threshold andhysteresis values of the codec modes is calculated by adding thehysteresis value to the lower threshold value of the initial codec mode.

BRIEF DESCRIPTION OF DRAWINGS

FIG. 1 illustrates a high level block diagram of a basic AMR speechsystem.

FIG. 2 is a flowchart describing a codec mode request under typical AMRconditions when there is an adequate filter state history for channelquality estimation.

FIG. 3 illustrates one process for initializing the filter state inorder to estimate channel quality.

FIG. 4 is an illustration of the filter state just after initializationusing the process of FIG. 3.

FIG. 5 illustrates another process for initializing the filter state inorder to estimate channel quality.

FIG. 6 is an illustration of the filter state after six channel qualityestimates have been obtained using the process of FIG. 5.

FIG. 7 identifies threshold and hysteresis points for each codec in anAMR system.

FIG. 8 illustrates the computed value and neighboring hysteresis methodsfor initializing the filter state.

FIG. 9 illustrates the computed value plus bias method for initializingthe filter state.

DETAILED DESCRIPTION

The present invention is concerned with determining a more reliable andaccurate initial channel quality estimate in an AMR system. Initialchannel quality estimate refers to the first estimate that occurs at thestart of a call or when a handoff to a new cell occurs. It is at theseinstances that there is no historical filter data to provide normalchannel quality estimates. Presently, there is no defined means fordetermining the quality of the channel at these instances.

There are several methods presented below for providing an initialchannel quality estimate. Before presenting the initial channel qualityestimation methods, it is useful to review a normal channel qualityestimate method. FIG. 2 is a flowchart describing a codec mode requestunder typical AMR conditions when there is an adequate filter statehistory for channel quality estimation. The first step is to determinean RF quality measurement using either bursts or frames 200. Next, a C/Ifilter is applied to the measurements 210. The currently used codec modeis then checked to see if it is the highest mode from the ACS 215. If itis, check 220 is skipped. If not, the filtered C/I value is thencompared to an upper threshold 220. If it exceeds the upper transitionpoint, then the next higher codec is requested 230. If it does notexceed the upper threshold, then it is next checked to see if thecurrent codec mode is the lowest from the ACS 225. If so, the same codecis requested 260. If not, the C/I value is checked against a lowerthreshold 240 provided the current codec mode is not the lowest in theACS. If the C/I measurement is lower than the lower threshold, then thenext lower codec is requested 250. Otherwise, if the C/I measurement iswithin the threshold tolerances, then the same codec is requested 260.

With respect to the process described above, the present invention isconcerned with step 270, C/I filtering.

The first initial channel quality estimate method is presented in FIG.3. It essentially initializes the entire filter state with the firstcalculated C/I value as determined from the first received burst orframe. This method has the advantage of forcing the channel qualityestimate to a steady-state value more quickly in most situations. Forsuch situations, this method avoids the problem of setting an initialnon-optimal codec when a new channel is assigned.

A received RF signal is analyzed to determine channel quality based onthe first burst 300. A C/I ratio for the first burst is calculated 310.Once calculated, this C/I ratio (e.g., 8 dB) is used to fill all thetaps in the filter state 320 and a codec is requested according to theprocess described in FIG. 2.

FIG. 4 is an illustration of the filter state just after initializationaccording to the process of FIG. 3. In this example, an 8 dB C/I ratiowas calculated and initialized into all 100 taps of the filter state. Acodec mode is then chosen based on this value. From this point forward,the filter state will be serially updated with actual values.

Initializing the filter with the first value may not always be optimal,however, since the variance of the channel quality measure can typicallybe quite high. This is particularly true in a faded environment and/orwhen frequency hopping is in use but may even be observed in a staticsingle-frequency channel due to randomness in the estimation technique.Hence, a received burst may be estimated to have quality considerablyhigher or lower than those received even immediately before or afterthat burst. The purpose of the long channel quality estimation filter isto smooth this randomness.

If the first estimated channel quality measure is atypical for thechannel, (i.e. an outlier), then the first calculated value method canlead to less than optimal results. The outlier will force a non-optimalcodec rate. Because it fills the entire buffer, this codec mode will beused for a considerable amount of time as the filter fills with newchannel estimates and the filtered value forces a change to a moreoptimal codec.

Another method for initializing the filter begins similarly to theprevious method where the filter is filled with the first calculatedvalue. However, subsequent channel quality estimates fill the filterstate proportionally. This is termed the proportional fill method. So,when available, the second channel quality estimate fills the secondhalf of the filter state leaving the first channel quality estimate inthe first half of the filter state. Likewise, the third channel qualityC/I estimate fills the last third of the filter state while the secondestimate fills the second third of the filter state and the firstestimate fills the first third of the filter state. This procedurecontinues until the filter is filled with values corresponding toseparate estimates at which point the filter would be run normally.

FIG. 5 illustrates the proportional fill process for initializing thefilter state in order to estimate channel quality when the underlyingC/I estimation is performed burst-wise. The current RF burst is measured500 for channel quality and a C/I ratio is calculated 510. If thecurrent burst is the first burst then the filter state is completelyfilled with this channel quality estimate 520. The process is repeatedfor each new burst. As new C/I ratios for new bursts are determined theyare proportionally filled into the filter state. FIG. 6 is anillustration of the filter state after six channel quality estimateshave been obtained using the process of FIG. 5. Thus, each C/I ratiorepresents ⅙^(th) of the filter state since, in our example, six channelquality measurements have been obtained. This is shown in FIG. 6 wherethe C/I values have been labeled from X₀ to X₅.

The proportional fill method recognizes that each new channel qualitymeasurement may create a situation where the total number of channelquality measurements is not evenly divisible into the number of taps inthe filter state. To accommodate such situations, the taps remainingafter dividing the total number of taps by the current number of channelquality measurements are filled with the most recent channel qualitymeasurement. For instance, a 100 tap filter that has a history of sevenchannel quality measurements would have each channel quality measurementoccupying 14 taps leaving two taps unaccounted. These last two tapswould be filled with the most recent channel quality measurement. Theproportional fill method can be represented mathematically as: Integer(Total number of taps/current number of channel quality measurements),with the remaining taps being filled with the most recent channelquality measurement.

Moreover, the proportional fill method applies until the number ofchannel quality measurements is equal to ½ the total number of filtertaps. When this threshold is reached, the filter is deemed to be insteady state and additional channel quality measurements are input tothe filter on a one in/one out basis.

One skilled in the art will recognize that an equivalent result may beobtained by altering the filter itself. To begin, no filter would beused. After the second estimate, a one-tap filter would be used and soforth.

The proportional fill method is advantageous over the first calculatedmethod in that it quickly finds the optimal codec in the typical casebut it also quickly reduces the effect of outlying channel qualityestimates. That is, the excessive weighting of the first estimate ismore quickly diminished.

Yet another method for determining initial channel quality is to computethe value used to initialize the C/I estimate filter state for every AMRcall and AMR handover using the ICM specified by the base station andappropriate threshold/hysteresis parameters. In considering the ICM, themobile takes advantage of the channel quality knowledge of the basestation.

FIG. 7 illustrates the usage of threshold and hysteresis points for eachcodec mode in an AMR system. Each codec mode spans a specified C/I rangeof values (in dB) that is associated with channel quality. A higher C/Ivalue indicates better RF channel quality and a lower higher C/I valueindicates poorer RF channel quality. Codec modes that span the lower C/Ivalues are configured to perform greater error correction for the speechsignal while codec modes that span the higher C/I values require lesserror correction. The C/I ranges for each codec mode are bounded by anupper and lower threshold value and also contain small C/I overlapsbetween adjacent codec modes. The overlap range is set by hysteresisparameters.

While FIG. 7 has been illustrated with four codec modes, there can befewer codec modes in a given AMR system. The number of codec modes doesnot affect the nature or operation of the present invention.

There can be many computational methods for initializing the filterstate. One method is to initialize the filter state with the mean valueof the upper and lower transition points corresponding to the ICMspecified by the base station.

Consider an example illustrated in FIG. 8 in which sample dB values havebeen applied to the codec mode threshold and hysteresis parameters. Inthe example, the base station specifies a fully provisioned ACS in whichcodec mode 3 is the ICM. From FIG. 8, it is known that the thresholdvalues for codec mode 3 are 10 dB (threshold 2 value) and 19 dB(threshold 3 value+hysteresis 3 value). The computed mean for codec mode3 is determined as (10 dB+19 dB)/2=14.5 dB. This is referred to as thecomputed value initialization method. Thus, all N taps of the filterstate are initialized with a 14.5 dB value.

Another approach to compute the initialization value for this situationinvolves offsetting the transition point by the hysteresis of theneighboring lower codec. Again, consider the threshold and hysteresisparameters in FIG. 8 with the ICM being codec mode 3. The transitionpoint from codec mode 2 to codec mode 3 is 10 dB and the hysteresisparameter for the overlap between codec modes 2 and 3 is 3 dB. Using theneighboring codec mode hysteresis offset method, the initial filterstate would be filled with a (10 dB+3 dB=13 dB) value.

Still another approach includes a combination of the aforementionedtechniques. Performance for the “Computed Value” method can be improvedby considering a limited number of actual C/I measurements. Several C/Imeasurements could be used to determine a crude C/I estimate. Knowledgeof actual channel quality allows the computed value to be furtherrefined before it is used to initialize the filter state.

The computed value could be improved by a limited knowledge of theactual RF channel quality. If the initial C/I measurement(s) suggest RFchannel quality is sufficient to support the next higher codec, theComputed Value is biased towards the ICM's upper transition point beforeinitializing the filter state. This results in fewer actual measurementsrequired before the mobile requests the next higher codec. Audio qualityfrom the user's perspective is improved since a higher speech bit ratecodec mode is requested sooner than with other methods.

An advantage is also realized for the situation where the crude C/Iestimate indicates RF channel quality is significantly less than theICM's lower transition point. By biasing the “computed value” downward,a lower speech bit rate will be requested sooner than with othermethods. Additional channel coding reduces dropped frames and thusimproves the user's overall experience. The bias up and bias downapproaches are illustrated in FIG. 9.

It is to be recognized by one skilled in the art that variouscombinations of the methods described above may also be used along withderivatives that are not explicitly discussed. The methods could beimplemented in software (DSP or general-purpose microprocessor),hardware, or a combination.

It should also be noted that the term “receiver” as used herein refersto the receiving portion of a cellular transceiving device. A cellulartransceiving device includes both a mobile terminal (MS) as well as abase station (BSS). A mobile terminal must be in communication with abase station in order to place or receive a call. There are numerousprotocols, standards, and speech codecs that can be used for wirelesscommunication between a mobile terminal and a base station.

Specific embodiments of the present invention are disclosed herein. Oneof ordinary skill in the art will readily recognize that the inventionmay have other applications in other environments. In fact, manyembodiments and implementations are possible. The following claims arein no way intended to limit the scope of the present invention to thespecific embodiments described above. In addition, any recitation of“means for” is intended to evoke a means-plus-function reading of anelement and a claim, whereas, any elements that do not specifically usethe recitation “means for” are not intended to be read asmeans-plus-function elements, even if the claim otherwise includes theword “means”.

1. For purposes of selecting an optimal codec mode in a multi-rate service, a method of estimating initial channel quality in a receiver when a new channel is allocated, said method comprising: obtaining channel quality measurements; calculating a carrier-to-interference (C/I) value of the obtained channel quality measurements; dynamically initializing a filter comprised of N taps with the calculated C/I values; and filtering the C/I values to remove randomness thereby providing a better channel quality estimation for use in codec mode selection within the multi-rate service; wherein the step of dynamically initializing the filter with the calculated C/I values comprises: obtaining the number of codec modes for the multi-rate service from a base station that is in communication with the receiver; obtaining threshold and hysteresis values for the codec modes from the base station; obtaining the initial codec mode from the base station; and calculating a C/I value to be used to initialize all N taps of the filter based upon the threshold and hysteresis values of the codec modes together with the initial codec mode.
 2. The method of claim 1 wherein the step of dynamically initializing the filter with the calculated C/I values comprises initializing all N taps of the filter with the first calculated C/I value.
 3. The method of claim 1 wherein the step of dynamically initializing the filter with the calculated C/I values comprises proportionally filling the N taps with each calculated C/I value.
 4. The method of claim 3 wherein each calculated C/I value occupies a number of taps equivalent to: Integer (n/current number of calculated C/I values), wherein any remaining taps are filled with the most recent calculated C/I value.
 5. The method of claim 1 wherein the C/I initialization value is calculated according to the formula: ((TH_(n)+H_(n))+TH_(n−1))/2, in which TH_(n) is the lower threshold value of the codec mode one higher than the initial codec mode; TH_(n−1) is the lower threshold value of the initial codec mode; and H_(n) is the hysteresis value for the overlap between the initial codec mode and the codec mode one higher than the initial codec mode.
 6. The method of claim 1 wherein, when the initial codec mode obtained from the base station is the lowest codec mode, then the value to be used to initialize all N taps of the filter is calculated according to the formula: (TH_(n)+TH_(LPL))/2, in which TH_(n) is the lower threshold value of the codec mode one higher than the initial codec mode; TH_(LPL) is the lowest practical threshold value of the lowest codec.
 7. The method of claim 1 wherein, when the initial codec mode obtained from the base station is the highest codec mode, then the value to be used to initialize all N taps of the filter is calculated according to the formula: TH_(HPL)+(TH_(n−1)+H_(n−1))/2, in which TH_(n−1) is the lower threshold value of the initial codec mode; H_(n−1) is the hysteresis value for the overlap between the initial codec mode and the codec mode one lower than the initial codec mode; and TH_(HPL) is the highest practical threshold value of the highest codec.
 8. The method of claim 1 wherein the C/I initialization value is calculated according to the formula: TH_(n−1)+H_(n−1), in which TH_(n−1) is the lower threshold value of the initial codec mode; and H_(n−1) is the hysteresis value for the overlap between the initial codec mode and the codec mode one lower than the initial codec mode.
 9. The method of claim 1 wherein, when the initial codec mode is the lowest codec mode, then the C/I initialization value is calculated according to the formula: TH_(n)−H_(n), in which TH_(n) is the lower threshold value of the codec mode one higher the initial codec mode; and H_(n) is the hysteresis value for the overlap between the initial codec mode and the codec mode one higher than the initial codec mode.
 10. The method of claim 1 wherein the filter is a finite impulse response (FIR) filter.
 11. The method of claim 10 wherein the number of filter taps is dynamically set to the number of received bursts up until a maximum filter size is reached.
 12. The method of claim 1 wherein the filter is an infinite impulse response (IIR) filter.
 13. The method of claim 1 wherein the C/I value calculated is biased toward a neighboring codec based on a number of the most recent channel quality measurements.
 14. For purposes of selecting an optimal codec mode in a multi-rate service, a system for estimating initial channel quality in a receiver when a new channel is allocated, said system comprising: means for obtaining channel quality measurements; means for calculating a carrier-to-interference (C/I) value of the obtained channel quality measurements; means for dynamically initializing a filter comprised of N taps with the calculated C/I values; and means for filtering the C/I values to remove randomness thereby providing a better channel quality estimation for use in codec mode selection within the multi-rate service; wherein the step of dynamically initializing the filter with the calculated C/I values comprises: means for obtaining the number of codec modes for the multi-rate service from a base station that is in communication with the receiver; means for obtaining threshold and hysteresis values for the codec modes from the base station; means for obtaining the initial codec mode from the base station; and means for calculating a C/I value to be used to initialize all N taps of the filter based upon the threshold and hysteresis values of the codec modes together with the initial codec mode.
 15. The system of claim 14 wherein the step of dynamically initializing the filter with the calculated C/I values comprises initializing all N taps of the filter with the first calculated C/I value.
 16. The system of claim 14 wherein the step of dynamically initializing the filter with the calculated C/I values comprises means for proportionally filling the N taps with each calculated C/I value.
 17. The system of claim 16 wherein each calculated C/I value occupies a number of taps equivalent to: Integer (N/current number of calculated C/I values), wherein any remaining taps are filled with the most recent calculated C/I value.
 18. The system of claim 14 wherein the C/I initialization value is calculated according to the formula: ((TH_(n)+H_(n))+TH_(n−1))/2, which TH_(n) is the lower threshold value of the codec mode one higher than the initial codec mode; TH_(n−1) is the lower threshold value of the initial codec mode; and H_(n) is the hysteresis value for the overlap between the initial codec mode and the codec mode one higher than the initial codec mode.
 19. The system of claim 14 wherein, when the initial codec mode obtained from the base station is the lowest codec mode, then the value to be used to initialize all N taps of the filter is calculated according to the formula: (TH_(n)+TH_(LPL))/2, in which TH_(n) is the lower threshold value of the codec mode one higher than the initial codec mode; TH_(LPL) is the lowest practical threshold value of the lowest codec.
 20. The system of claim 14 wherein, when the initial codec mode obtained from the base station is the highest codec mode, then the value to be used to initialize all N taps of the filter is calculated according to the formula: T_(HPL)+(TH_(n−1)+H_(n−1)))/2, in which TH_(n−1) is the lower threshold value of the initial codec mode; H_(n−1) is the hysteresis value for the overlap between the initial codec mode and the codec mode one lower than the initial codec mode; and TH_(HPL) is the highest practical threshold value of the highest codec.
 21. The system of claim 14 wherein the C/I initialization value is calculated according to the formula: TH_(n−1)+H_(n−1), in which TH_(n−1) is the lower threshold value of the initial codec mode; and H_(n−1) is the hysteresis value for the overlap between the initial codec mode and the codec mode one lower than the initial codec mode.
 22. The system of claim 14 wherein, when the initial codec mode is the lowest codec mode, then the C/I initialization value is calculated according to the formula: TH_(n)−H_(n), in which TH_(n) is the lower threshold value of the codec mode one higher the initial codec mode; and H_(n) is the hysteresis value for the overlap between the initial codec mode and the codec mode one higher than the initial codec mode.
 23. The system of claim 14 wherein the filter is a finite impulse response (FIR) filter.
 24. The system of claim 23 wherein the number of filter taps is dynamically set to the number of received bursts up until a maximum filter size is reached.
 25. The system of claim 14 wherein the filter is an infinite impulse response (IIR) filter.
 26. The system of claim 14 wherein wherein the C/I value calculated is biased toward a neighboring codec based on a number of the most recent channel quality measurements. 